Voice over Internet Protocol (VoIP) is a general term for a family of transmission technologies used to deliver voice communications over IP networks such as the Internet or other packet-switched networks. Other terms frequently encountered and synonymous with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone.
Internet telephony refers to communications services—voice, facsimile, and/or voice-messaging applications—that are transported via the Internet, rather than the public switched telephone network (PSTN). The basic steps involved in originating an Internet telephone call include conversion of the analog voice signal to digital format and translation of the signal into Internet protocol (IP) packets for transmission over the Internet; the process is reversed at the receiving end.
VoIP systems employ session control protocols, such as the Session Initiation Protocol (SIP), to control the set-up and tear-down of calls as well as audio codecs which encode speech allowing transmission over an IP network as digital audio via an audio stream. The advantage to VoIP is that a single network can be utilized to transmit data packets as well as voice and video packets between users, thereby greatly simplifying communications.
SIP is an open signaling protocol for establishing many kinds of real-time and near-real-time communication sessions, which may also be referred to as dialogs. Examples of the types of communication sessions that may be established using SIP include voice, video, and/or instant messaging. These communication sessions may be carried out on any type of communication device such as a personal computer, laptop computer, telephone, cellular phone, Personal Digital Assistant, etc. One key feature of SIP is its ability to use an end-user's Address of Record (AOR) as a single unifying public address for all communications. Thus, in a world of SIP-enhanced communications, a user's AOR becomes their single address that links the user to all of the communication devices associated with the user. Using this AOR, a caller can reach any one of the user's communication devices, also referred to as User Agents (UAs) without having to know each of the unique device addresses or phone numbers.
IETF SIP standards, specifically, RFC 4028 describes a SIP signaling session refresh technique for monitoring the health of an established call session. The technique described in RFC 4028 requires SIP signaling peers to exchange periodic SIP in-dialog INVITE or UPDATE messages to check and extend the status of the SIP dialog. However, it makes a simplifying recommendation by tying the status of the media connection to the status of the SIP dialog. Specifically, if the SIP dialog refresh times out or fails (by receipt of a non 2xx final response message), the media associated with the SIP dialog is automatically terminated. While this approach may be desirable behavior for example, to prevent billing errors, it results in immediate media loss, which is undesirable in many circumstances. Since the path of SIP Signaling often does not follow the same path as the connection media, it is possible to preserve the media connection even though the end-to-end SIP signaling connection has been disrupted.